Summary:	ASTERISK-26689: res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting channel for lack of RTP activity
Reporter:	Dmitriy Serov (Demon)	Labels:
Date Opened:	2017-01-03 12:21:16.000-0600	Date Closed:	2020-01-14 11:14:06.000-0600
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Resources/res_pjsip_sdp_rtp
Versions:	13.7.0	Frequency of Occurrence	Constant
Related Issues:
Environment:		Attachments:	( 0) 183-lack-rtp.txt
Description:	Asterisk 13.7 (chan_pjsip) & Zyxel Keenetic Plus DECT, firmware v2.06(AAGJ.9)C1 Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip trunk). Call using early media (183 Session in progress) and rtp_timeout=10. After 10 seconds: [2016-12-16 13:53:15] NOTICE[6654] res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/xxx-0000027b' for lack of RTP activity in 10 seconds SIP dump is attached. According to [1] before called user agent send OK or ACK there is one way SDP. In sip dump (attached) i didn't find such SIP packets. Whether that call was canceled due to RTP inactivity?
Comments:	By: Asterisk Team (asteriskteam) 2017-01-03 12:21:17.524-0600 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. By: Dmitriy Serov (Demon) 2017-01-03 12:22:32.101-0600 sip dump of call with lack of rtp activity. By: Rusty Newton (rnewton) 2017-01-05 09:16:12.514-0600 Pretty sure this was fixed recently, or something very similar. I'm trying to find it, but in the meantime can you test with the latest of the 13 branch? By: Asterisk Team (asteriskteam) 2017-01-19 12:00:01.219-0600 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines By: Friendly Automation (friendly-automation) 2022-03-25 17:26:46.688-0500 Change 18071 merged by Friendly Automation: res_pjsip_sdp_rtp: Improve detecting of lack of RTP activity [https://gerrit.asterisk.org/c/asterisk/+/18071|https://gerrit.asterisk.org/c/asterisk/+/18071] By: Friendly Automation (friendly-automation) 2022-03-25 17:30:48.238-0500 Change 18232 merged by Friendly Automation: res_pjsip_sdp_rtp: Improve detecting of lack of RTP activity [https://gerrit.asterisk.org/c/asterisk/+/18232|https://gerrit.asterisk.org/c/asterisk/+/18232] By: Friendly Automation (friendly-automation) 2022-04-06 04:02:56.120-0500 Change 18231 merged by Joshua Colp: res_pjsip_sdp_rtp: Improve detecting of lack of RTP activity [https://gerrit.asterisk.org/c/asterisk/+/18231|https://gerrit.asterisk.org/c/asterisk/+/18231] By: Friendly Automation (friendly-automation) 2022-04-06 04:03:19.702-0500 Change 18230 merged by Joshua Colp: res_pjsip_sdp_rtp: Improve detecting of lack of RTP activity [https://gerrit.asterisk.org/c/asterisk/+/18230|https://gerrit.asterisk.org/c/asterisk/+/18230]